对于pcm8格式的语音压缩16bit的声音windowpcm，压缩率…

语音编码标准ｇ．７２３的压缩源码，ＶＣ上编译通过，可参考-Voice coding standard g. 723 compression source, VC compiled through reference

这是源代码 IIR，它使用学调节效应，筛选器音频，它为使用 DSP，它通常为 DSK 6416 和它通常应用研究上大学科技、 行业

会转化成声波。

gsm 标准参考代码-GSM standard reference source

windows SDK中的音频处理样列代码，主要使用WASAPI来抓取音频数据，学习音频的好资料-

来源于网上的C++技术开发的音频处理代码, 为国外开源代码, 可用来学习掌握C++的MP3音频处理算法。来源于网上公开资料，供学习使用。-From online C++ Technology development of audio processing code, for foreign open-source code, can be used to study and master the C++ The MP3 audio processing algorithms. Disclosure of information from the Internet for learning to use.

i writing an free losless audio coder, faster on old 486 than flac or ape decoding on pentium 233 MHz !!! the program working, and compression ratio is equal to flac / ape ... On pentium 233 MHz flac decompress time is 26 sec of enya_.wav . ape is more than 45 sec... :) approx 14 sec coding enya.wav with pkzip, and my routines not have completed. My hope is 4 sec compression time, if completing, and decompressing a 16 bit stereo 44100 Hz is real time on 80286 cpu !!! PLEASE HELP ME ! i am an hungarian programmer, and i cannot wrtite in english perfectly. i not have time for write code for incoming 16 bit variable contents ! i want variable bit size putting into 8 bit foam. the 16 bit variable containing e.g. 7 bit valuable data, and 8 of this data must be storing into 8 bit datas. (code need 24 .. 2 bit incoming packing into 8 bit data.)-i writing an free losless audio coder, faster on old 486 than flac or ape dec

又一种新的97小波的C代码实现 可以参考一下-Also a new 97 wavelet C code can refer to realize

java录音放音-Java record/play voice

应用背景在利用codec进行语音数据输入和发送时，可以利用该文件提供的函数对输入和输出的pwm语音数据的音量进行缩放。关键技术对PWM语音数据进行幅值上的缩放，从而达到对codec输入或者输出的pwm语音数据进行缩放的目的。控制的参数是对语音数据进行控制的系数，0值语音数据音量就为0了。